Acoustic echo cancellers are used in amplifying call apparatuses that perform transmission and reception of voice through, for example, a transmission path, amplify and output, through a loudspeaker, the voice (far-end voice) received from a destination terminal, and transmit the voice (near-end voice) picked by a microphone. If far-end voice output through the loudspeaker is picked up by the microphone, a phenomenon will occur in which the voice emitted from the destination terminal returns thereto as an echo (acoustic echo). In general, since a certain delay occurs in the transmission path, the acoustic echo may well disturb voice communication.
To cancel or reduce such an acoustic echo, echo cancellers are used. At an originator terminal, a received input as the cause of the acoustic echo is detected. If a transformation corresponding to the transfer characteristic of an echo path is executed on the received input, a replica of an acoustic echo mixing in the output of the microphone can be produced internally. By subtracting the echo replica from the microphone output, an output signal (error signal) with suppressed acoustic echo can be produced. The mechanism for achieving this process is an echo canceller.
In an echo canceller, an acyclic linear filter of a tap number N (FIR filter) is often used as a linear filter for simulating the transfer characteristic of the echo path. By performing convolution of the tap coefficient of the filter and the received input, an echo replica is produced.
In general, it is difficult to instantly and correctly impart, to a filter, the transfer characteristic of the echo path that varies instantaneously. Therefore, an adaptive algorithm for asymptotically detecting the transfer characteristic based on an observed signal is utilized. As a group of adaptive algorithms, stochastic gradient algorithms are known in which the tap coefficient is corrected in accordance with an instantaneous square error gradient (stochastic gradient) associated with the tap coefficient. As the stochastic gradient algorithms, a least mean square error (LMS) algorithm or a normalized LMS (NLMS) algorithm are known.
However, the tap coefficient correcting amount in an LMS algorithm or NLMS algorithm is proportional to a reference signal and an error signal, and is therefore very great. Because of this, during double talk in which near-end voice and far-end voice coexist, significantly erroneous adjustment may occur. To avoid such erroneous adjustment, it is necessary to suppress the tap coefficient correcting amount or to completely stop the correction during double talk. When employing such algorithms, it is necessary to provide a double-talk detector for detecting double talk to control a tap coefficient correcting unit.
Japanese Patent No. 3870861 (Patent Document 1) discloses an echo canceller utilizing an independent component analysis (Infomax formula) based on information entropy maximization, in which canceller, a tap coefficient is corrected so as to make a reference signal and an error signal independent of each other. The echo canceller of Patent Document 1 employs a tap coefficient correcting expression wherein the function G(e(t)) of an error signal e(t) is set to a sign function: sign(e(t)), a hyperbolic tangent function: tan h(e(t)), or a sigmoid function: 1/(1+exp(−e(t)). Algorithms using these functions will now be collectively referred to as “the Infomax formula.” In the Infomax formula, the coefficient correction scale does not exceed a preset peak level, which is considered to suppress erroneous correction during double talk. Because of this, the Infomax formula does not require a double talk detector and therefore enables the entire system to be made more compact.
However, in the Infomax formula, the residual echo level is high, i.e., echo cancellation is insufficient. This may be because the scale of coefficient correction performed for a minimal-error is excessively large and hence overshoot will occur.
Japanese Patent No. 2885269 (Patent Document 2) discloses a tap coefficient correcting expression, the function G(e(t)) of which includes a linear zone wherein the amount of correction is proportional to the error signal e(t) in a small error region in a sign algorithm. In the algorithm of Patent Document 2, the overshoot suppression effect of the linear zone realizes a residual echo level lower than in the sign algorithm. In other words, the algorithm of Patent Document 2 is an algorithm obtained by providing the LMS algorithm with a correction scale limiter. This algorithm will be hereinafter referred to as “the Ideal Limiter formula.” By virtue of the limiter effect, the algorithm of Patent Document 2 exhibits robustness against double talk, like the Infomax formula. However, the convergence speed of the tap coefficient in the Ideal Limiter formula is lower than in the Infomax formula. It is apparent that the reason for it lies in the relative small coefficient correction scale in the linear zone.
JP-A 2004-64681 (KOKAI) (Patent Document 3) discloses an adaptive algorithm that provides an effect of accelerating the same convergence as a least mean fourth (LMF) algorithm. However, the algorithm of Patent Document 3 does not exhibit robustness against double talk, which is similar to the LMS algorithm. Further, in this algorithm, since the tap coefficient correction scale monotonically increases in accordance with the instantaneous absolute value of an error signal, it exhibits a significant error correction during double talk, as in the LMF algorithm.